1. Field of the Invention
The present invention relates to MPEG audio bitstream encoding/decoding, and more particularly, to a method of and an apparatus for encoding/decoding an MPEG-4 bit sliced arithmetic coding (BSAC) audio bitstream having ancillary information.
2. Description of the Related Art
An analog waveform is a continuous-time signal. Therefore, analog-to-digital (A/D) conversion is necessary to represent the analog waveform as a discrete-time signal. Two processes are necessary for the A/D conversion. One is a sampling process for converting a temporally continuous-time signal into a discrete-time signal, and the other is an amplitude quantization process for limiting the number of possible amplitudes using a finite value. That is, the amplitude quantization process converts an input amplitude x(n) at a time n to y(n), which is an element of a finite set of possible amplitudes.
In an audio signal storing/restoring method, according to recent development of digital signal processing technologies, a technology of sampling and quantizing a typical analog signal, converting the sampled and quantized signal to pulse code modulation (PCM) data, which is a digital signal, storing the PCM data in a recording/storing medium such as a compact disc (CD) or a digital audio tape (DAT), and listening to the PCM data by reproducing the stored data according to a user demand has been developed. By applying the storing/restoring method using a digital method, better sound quality may be obtained and deterioration due to a stored duration may be prevented as compared with tape recording using an analog method such as a long-play record (LP). However, since a size of digital data is great, problems occur when storing or transmitting is performed.
To solve the storage and transmission problems, efforts to reduce data amount using a differential pulse code modulation (DPCM) method or an adaptive differential pulse code modulation (ADPCM) method, which compresses a digital voice signal, are being made. However, efficiency in the DPCM or ADPCM method is largely different according to the kinds of signals. Recently, in Moving Picture Expert Group (MPEG)/audio technologies for which standardization works have been achieved by International Standard Organization (ISO) or AC-2/AC-3 technologies developed by DOLBY CO. LTD., a method of reducing data amount by using a psychoacoustic model has been used. The method of reducing the data amount has largely contributed to efficiently reducing data amount regardless of signal characteristics.
In a conventional audio compression technology such as MPEG-1/audio, MPEG-2/audio, or AC-2/AC-3, signals in the time domain are bound in blocks having a predetermined size and converted to signals in the frequency domain. The converted signals are scalar quantized using a psychoacoustic model. The quantizing technology is simple but not optimum even if an input sample is statistically independent. Furthermore, if the input sample is statistically dependent, the quantizing technology is inefficient. Due to this problem, encoding is performed by including lossless encoding, such as entropy encoding, or a certain kind of adaptive quantization. Therefore, a more complicated process than storing simple PCM data is performed, and a bitstream is composed of quantized PCM data and ancillary information for signal compression.
The MPEG/audio standard or AC-2/AC-3 method provides sound quality equivalent to the sound quality of a CD with a 64 Kbps-384 Kbps rate, which is a ⅙ to ⅛ of a conventional digital encoding rate. With high sound quality, the MPEG/audio standard will play an important role for an audio signal storing and transmitting system such as digital audio broadcasting (DAB), an internet phone, audio on demand (AOD), or a multimedia system.
In conventional methods, since a fixed bitrate is provided in an encoder and a quantizing and encoding process is performed by finding an optimal status for the provided bitrate, when a fixed bitrate is used for encoding, the methods provide a good scheme. However, for multimedia purposes, there is a need for conventional low bitrate encoding and encoders/decoders having various functions. One of these is an audio encoder/decoder capable of controlling a bitrate. The bitrate controllable audio encoder can make a low bitrate bitstream using a bitstream encoded with a high bitrate and restore the bitstream using only a partial bitstream. Accordingly, when a network is overloaded, when a performance of a decoder is not good, or when a bitrate is lowered by a user's demand, the bitrate controllable audio encoder should restore an audio signal with a reasonable performance using a partial bitstream even though the performance is deteriorated by the lowered bitrate.
A syntax allowing ancillary information to be stored, such as data_stream_element( ) and fill_element( ), is in the MPEG-2/4 AAC (ISO/IEC 13818-7, ISO/IEC 14496-3). Also, “ancillary data” is defined in the MPEG-1 layer-III (mp3). Accordingly, audio ancillary information may be stored by embedding the ancillary information in the middle of frame information. ID3v1 is a representative example in this respect. FIG. 11 shows a bitstream structure of ID3v1.
However, a syntax allowing ancillary information to be provided is not defined in a currently standardized MPEG-4 bit sliced arithmetic coding (BSAC) audio format. FIGS. 12 and 13 show a definition of a frame header of a BSAC syntax. In the BSAC, since a syntax allowing ancillary information to be embedded is not defined in a frame header, according to the standard, it is impossible to embed the ancillary information in the frame header.